[..] |
ASTERISK-02000: [post-1.0][patch] Custom command line for musiconhold software execution |
ASTERISK-02001: AGI script does not see first digit from SIP cahnnel (using DTMF via SIP INFO) |
ASTERISK-02002: Zaptel linux26 compile looks for /usr/src/linux-2.6/arch/x86 instead of /arch/i386 |
ASTERISK-02003: mobile: number not assigned/tmp. unavail: gives no event |
ASTERISK-02004: Problem with Sip UNKN (d) |
ASTERISK-02005: [patch] Notify on EOL of sip.conf incominglimit/outgoinglimit |
ASTERISK-02006: [patch] Enhancement to "ParkedCalls" action in astman |
ASTERISK-02007: cannot transfer calls placed on PSTN trunk |
ASTERISK-02008: [patches][src-audit] apps directory files app_a*.c through app_m*.c |
ASTERISK-02009: [PATCH] Teach RPM not to replace config files that already exist |
ASTERISK-02010: [CONTRIB] Prepaid billing and rating apps |
ASTERISK-02011: [patches][src-audit] apps directory files app_q*.c through app_z*.c |
ASTERISK-02012: [patches][src-audit] directories: astman, cdr, db1-ast, editline, pbx, res, and stdtime |
ASTERISK-02013: [patch] res_parking.c missing an include--get compiler warning |
ASTERISK-02014: Strange behavior of zaptel T100p |
ASTERISK-02015: RFC2833 + app_senddtmf |
ASTERISK-02016: [patch] Different dialplan for caller number |
ASTERISK-02017: RFC2833 + app_senddtmf |
ASTERISK-02018: RFC2833 + app_senddtmf |
ASTERISK-02019: ASTCC permission problem |
ASTERISK-02020: Problems with insecure= setting and authentication |
ASTERISK-02021: [patch][src-audit] channels directory -- last one |
ASTERISK-02022: Coredump when waiting stream |
ASTERISK-02023: [post-1.0][patch] Added Support to map table and cols in conf files for IAX2, SIP and VM |
ASTERISK-02024: [patch] pbx.c: Code formatting and help text updates |
ASTERISK-02025: [patch] assign more group for any channel (app_groupcount) |
ASTERISK-02026: [patch] SIP registers outbound at wrong port |
ASTERISK-02027: [post-1.0][patch] Sending HTML voicemail messages |
ASTERISK-02028: [patch] app_voicemail is unable to update config in res_config_odbc |
ASTERISK-02029: [PATCH] Tell RPM to put header files in a separate -devel.rpm |
ASTERISK-02030: [patch] Update of ast_expr to use flex scanner - spaces no longer necc. |
ASTERISK-02031: asterisk.c - not releasing allocated CLI memory |
ASTERISK-02032: SIP SUBSCRIBE - NOTIFY Wrong Call-ID Match |
ASTERISK-02033: mgcp reload stops asterisk from monitoring mgcp socket |
ASTERISK-02034: Really long first ring, then normal |
ASTERISK-02035: [patch] Let chan_zap compile when there is no libpri |
ASTERISK-02036: [patch] Makefile error for chan_h323 linking |
ASTERISK-02037: [patch] 7960 rejecting from field with :0 port description |
ASTERISK-02038: [patch] Queue: new event when ringing phone, before bridging |
ASTERISK-02039: [FreeBSD Only] Coredump in var resolving/substituting |
ASTERISK-02040: bad checksum udp packets on linux |
ASTERISK-02041: [patch] Some enhancements for astcc |
ASTERISK-02042: externnotify script is always called even with no voicemail messages |
ASTERISK-02043: [Request] Voicemail subscriber cannot send a new message |
ASTERISK-02044: generating unique string for sending emails |
ASTERISK-02045: /proc/zaptel/xxx doesn't work properly under Linux 2.6 |
ASTERISK-02046: pri channels lock after a few days, asterisk restart doesn't clear, needs zaptel restart |
ASTERISK-02047: [patch] pbx.c: add skip and noanswer option to Background command |
ASTERISK-02048: ast_dtmf_stream fails to return correct values |
ASTERISK-02049: [post-1.0] [patch] An application that will help a user setup their voicemail account |
ASTERISK-02050: [request] Privacy flag in SetCallerID, SetCIDNum, SetCIDName |
ASTERISK-02051: [patch] RedHat asterisk init script replacement |
ASTERISK-02052: [patch] res/Makefile contains line referencing parking.h which has been renamed |
ASTERISK-02053: Missing DNS SRV support for SIP registration |
ASTERISK-02054: SIP generated inband DTMF too short |
ASTERISK-02055: SIP Re-invite still broken on some phones |
ASTERISK-02056: callerid restriction list |
ASTERISK-02057: [patch] Add command line arguments to safe_asterisk |
ASTERISK-02058: [patch] chan_zap.c compiler warning / possible wrong value to libpri function |
ASTERISK-02059: dsp.c fails to compile |
ASTERISK-02060: dsp.c fails to compile |
ASTERISK-02061: RTP.c not compiling on FreeBSD |
ASTERISK-02062: MGCP wildcard endpoint audit fails |
ASTERISK-02063: [Design & patch] app_privacy & confused users |
ASTERISK-02064: [patch] Two minor Makefile-related patched and a useful addition to frame.c |
ASTERISK-02065: Over amplified echo at start of call |
ASTERISK-02066: [patch] Allow rejected VoIP dial to return to dialplan |
ASTERISK-02067: [patch] Add return value check of ast_smoother_feed() before ast_smoother_read() |
ASTERISK-02068: [patch] Updated help text for dial() |
ASTERISK-02069: Improved counters to aid T1/E1 troubleshooting |
ASTERISK-02070: Tiny patch to make app_dial return "CANCEL" when user press * |
ASTERISK-02071: Changing useragent requires a restart |
ASTERISK-02072: Registering with an intertex ix66 no longer works |
ASTERISK-02073: Kernel Panic in Linux 2.6 when connection is broken to asterisk server |
ASTERISK-02074: [patch] Missing return in chan_h323, function connection_made |
ASTERISK-02075: Patch to clean many warnings in compilation of chan_h323 |
ASTERISK-02076: [patch] 'Newexten' manager event additional information |
ASTERISK-02077: Zaptel or Libpri: PRI protocol error reseting calls |
ASTERISK-02078: iaxy with one-way audio |
ASTERISK-02079: Avoided deadlock |
ASTERISK-02080: E1 stopped working in recent CVS. |
ASTERISK-02081: Installing Asterisk from CVS on YDL 3.0.1 error |
ASTERISK-02082: [FreeBSD 5.2.1. Only] Music On Hold fork blocks other threads |
ASTERISK-02083: iaxy loses registrations |
ASTERISK-02084: [*BSD only?] Invalid poll() processing, channel.c; Packets with bad UDP checksum. |
ASTERISK-02085: [patch] simple ast_log()-like debugging |
ASTERISK-02086: In build after 7/06/04 MWI on ADSI phones do not light up. But get shuttle tone |
ASTERISK-02087: [post-1.0] [patch] Allow Caller TON to be retrieved in the dialplan |
ASTERISK-02088: [patch] Remove quotation marks around MD5 arg in digest auth |
ASTERISK-02089: [Patch] add manager events in sip and IAX. Back port from chan_sip2x.c. |
ASTERISK-02090: [patch] [post 1.0] Improved SIP friends, supported postgres |
ASTERISK-02091: [patch] Make astman compile on FreeBSD 4.9 |
ASTERISK-02092: [patch] memset on possible NULL pointer |
ASTERISK-02093: [request] carrier grade CDR features requested |
ASTERISK-02094: Dlink doesn't provide tone when handset is picked up |
ASTERISK-02095: [Post-1.0][patch] Allow adding/removing queue members via manager API |
ASTERISK-02096: X100P driver crashes 4kstacks kernel |
ASTERISK-02097: [patch] add/remove members to/from queues on cli |
ASTERISK-02098: [patch] new cdr module using asterisk manager |
ASTERISK-02099: G.726 endianness not RFC compliant |
ASTERISK-02100: DTMF in voicemail not completely recognized |
ASTERISK-02101: [patch] Adding Belgium tones data |
ASTERISK-02102: [patch] debug peer port correction |
ASTERISK-02103: [patch] Port to Darwin 6.8/ OS X 10.3 |
ASTERISK-02104: Caller ID on FXS channels with Distinctive Ring |
ASTERISK-02105: SIP Cancels Fail due to Request-URI Mismatch |
ASTERISK-02106: [patch] ADSIProg() fails on exactly 245 bytes |
ASTERISK-02107: outgoing sip calls ACK timeout error (wrong call-id) NAT server |
ASTERISK-02108: chan_h323 & h.323 trace 2 and higher |
ASTERISK-02109: ast_verbose() duplicate log messages if there is no "\n" at the end of the string. |
ASTERISK-02110: DTMF stops working in Voicemail |
ASTERISK-02111: [post-1.0] [request] MGCP Support As MG/SG, not CA/MG/SG |
ASTERISK-02112: [patch] fxshonormode fix |
ASTERISK-02113: [request] Pager e-mail from field |
ASTERISK-02114: When using Meetme(<conf #>), won't retry if you enter incorrect pin |
ASTERISK-02115: [new app] [post-1.0] app_sql_mysql |
ASTERISK-02116: [patch] chan_alsa doesn't transmit, ignores configuration |
ASTERISK-02117: [patch] Fail to get redirecting number |
ASTERISK-02118: [patch] fix date values |
ASTERISK-02119: Dial always goes to busy, never to unavailable |
ASTERISK-02120: Add DNID to CDR |
ASTERISK-02121: [request] Add ast_dsp to generate manager events for talk/silence. |
ASTERISK-02122: [request] Be able to play audio files to an entire conference or specific user. |
ASTERISK-02123: [request] Exit Meetme on ANY DTMF key |
ASTERISK-02124: [request] Dispatch to extension based on *[0-9] |
ASTERISK-02125: [request] Play audio file in app_meetme if user gets voicemail |
ASTERISK-02126: [request] Specify ZapBarge audio direction |
ASTERISK-02127: [request] ZapBarge - Ability to change channel without exiting zapbarge |
ASTERISK-02128: [patch] remove option "outgoinglimit" that doesn't work anyway |
ASTERISK-02129: issues with wrapuptime |
ASTERISK-02130: RTP stream sent from wrong IP address if SIP address set to secondary IP address. |
ASTERISK-02131: IAX phones cannot transfer some outgoing channels |
ASTERISK-02132: [patch] Manager command ZapDialOffhook crashes when channel doesn't exist |
ASTERISK-02133: Remove the cdr_pgsql.conf + Asterisk reload = Crash (Segmentation fault) |
ASTERISK-02134: [patch] Add username to "sip show peer" |
ASTERISK-02135: Slash operator to match extension based on callerid does not play well with changing callerid |
ASTERISK-02136: fax detect does not detect fax from Brother Intellafax 4100 |
ASTERISK-02137: Modified some calls to permit functionality with Microsoft SQL Server |
ASTERISK-02138: [patch] MOH pthread rather than fork |
ASTERISK-02139: [request + patch] have call queues using a SIP member honor a 302 redirect |
ASTERISK-02140: Cisco 7960 SIP incoming call issue |
ASTERISK-02141: [patch] Show manager command required privileges in output of 'show manager commands' |
ASTERISK-02142: [patch] Adding and removing agents with manager command should require Agent privs |
ASTERISK-02143: Zap channel permanantly in conference after transferring to meetme |
ASTERISK-02144: [patch] Allow priority to be set even if non-root user ID is specified |
ASTERISK-02145: [patch] Allow the command-line editor to be chosen |
ASTERISK-02146: [patch] Prevent blank lines from being saved in the history |
ASTERISK-02147: Changes in Translation Path |
ASTERISK-02148: [patch] Exit cleanly from remote control mode with SIGINT etc. |
ASTERISK-02149: [patch] Default username from peer entry when creating new IAX2 channel |
ASTERISK-02150: [PATCH] H.323 Memory corruptions |
ASTERISK-02151: Limit number of calls to Agent |
ASTERISK-02152: chan_mgcp.c does not support PING event |
ASTERISK-02153: correct way to set wildcard endpoint in mgcp.conf |
ASTERISK-02154: [request] res_musiconhold be reload enabled |
ASTERISK-02155: [patch] allows specification of digittimeout when dialing extension during tranfers |
ASTERISK-02156: patch from 2174 wrong fix |
ASTERISK-02157: [patch] retrieve zapata channel status through asterisk manager |
ASTERISK-02158: Uncompressed version of asterisk/sounds and asterisk-sounds/sounds |
ASTERISK-02159: "mgcp reload" VERY buggy when mgcp.conf has wcardep= defigned |
ASTERISK-02160: LOCAL_USER_ADD |
ASTERISK-02161: translate.h using c++ |
ASTERISK-02162: ast_rtp_senddigit is hardcoded to use payload type 101 |
ASTERISK-02163: DTMF to SIP channel sent too soon |
ASTERISK-02164: [patch] Fix compile failure on OpenBSD 3.5 (Release) |
ASTERISK-02165: cdr_tds won't compile with older versions of FreeTDS |
ASTERISK-02166: [report + patch] Forwarding issue with multiple dial targets |
ASTERISK-02167: cdr_sqlite.so fail to create cdr.db+table |
ASTERISK-02168: Asterisk does not hang up SIP call |
ASTERISK-02169: inet_addr is not good to find out gw name is dotted ip or not ... |
ASTERISK-02170: 'mgcp reload' always reloads wildcard endpoints reguardless if they were loaded on startup |
ASTERISK-02171: When making a dynamic conference, no protection against duplicate conference numbers |
ASTERISK-02172: failed to reset conferencing |
ASTERISK-02173: Fix for bug #2200 doesn't check for dynamic_pin being null pointer |
ASTERISK-02174: [patch] Add user number to MeetmeEnter and MeetmeLeave events |
ASTERISK-02175: [post-1.0] [patch] h323-to-h323 and h323-to-mgcp native bridging |
ASTERISK-02176: [docs patch] cdr documentation, well a start |
ASTERISK-02177: No Ringback for ingress PSTN calls |
ASTERISK-02178: [patch] add call timer on cli show channel and manager status |
ASTERISK-02179: [patch] introduce the 'f' flag in app_directory (Directory()) to use the first name as a match |
ASTERISK-02180: [patch] Make the ADSI feature download number and security code configurable |
ASTERISK-02181: [patch] bugfix: asterisk may send SIP UA codec not offered in INVITE |
ASTERISK-02182: [patch] Announce parking extension on ADSI compatable CPE |
ASTERISK-02183: Return expression for wav_tell() wrong in format_wav_gsm.c |
ASTERISK-02184: [patch] Verbose app |
ASTERISK-02185: SIGBUS on sparc64/Linux |
ASTERISK-02186: No ringing when I dial |
ASTERISK-02187: [patch] * does not send back 'request identifier' in 'request notify' when wildcard endpoint sends 'notify' |
ASTERISK-02188: [patch] Fix verbose output to not ignore replace parameter |
ASTERISK-02189: wcfxs not detecting incoming calls |
ASTERISK-02190: bug in dsp.c with silence detection |
ASTERISK-02191: Avoid deadlock (2) |
ASTERISK-02192: [patch] Avoid duplicate IP addresses/registrations in sip_friends database |
ASTERISK-02193: call limit L(...) reset to initial value everytime something is dialed. |
ASTERISK-02194: [patch] From, To, etc. messages do not handle < in quoted-strings |
ASTERISK-02195: [patch] Asterisk sends corrupt data when peer dynamically switches from GSM to ULAW |
ASTERISK-02196: Dead Lock on IAX2 start from 01 August to today using IaxComm to call |
ASTERISK-02197: If videosupport=yes, SDP response on "voice only" calls include TWO media streams in SDP (one for voice and one for video) |
ASTERISK-02198: Internal clocking on a 2nd of 2 TE410P boards |
ASTERISK-02199: problem with Feature Group D / E&M Wink |
ASTERISK-02200: Tab Completion Repeating |
ASTERISK-02201: [Patch] correct MeetMe for Marked users |
ASTERISK-02202: Please provide PRI_HANGUPCAUSE |
ASTERISK-02203: Crash on CVS |
ASTERISK-02204: [patch] Allow use of on/off in ast_true() for validation of config options |
ASTERISK-02205: [patch] create dynamically conference with the CLI |
ASTERISK-02206: negative lag causes jitter buffer to grow larger than maxexcessjitter setting |
ASTERISK-02207: [request] make generators work even if no received audio available |
ASTERISK-02208: SNDCTL_DSP_SETDUPLEX does not work |
ASTERISK-02209: Extension 'T' (AbsoluteTiemout) inside a macro does not work as expected. |
ASTERISK-02210: ADSI voicemail folders softkeys display incorrectly |
ASTERISK-02211: [patch] [post-1.0] allow TON, NPI and Presentation to be retrieved in the dialplan |
ASTERISK-02212: [patch] MGCP does not currently have support for pre-rfc mode "draft" operation or future rfc definitions |
ASTERISK-02213: [patch] manager getvar action missing double CRLF in Response |
ASTERISK-02214: [patch] Add ";user=phone" when INVITE contain only phone number |
ASTERISK-02215: [report + patch] System() application troubles. |
ASTERISK-02216: Please include indicator for in/outbound in CDR-table |
ASTERISK-02217: MGCP call pickup with *8 and *8x options |
ASTERISK-02218: Asterisk doubles DTMF events when endpoint in INBAND mode |
ASTERISK-02219: [patch] [h.323] Auto dialing crashes |
ASTERISK-02220: Channel variables in AGI application after Dial. |
ASTERISK-02221: [patch] Implement fax detection for SIP calls. |
ASTERISK-02222: chan_h323 memory leak ? |
ASTERISK-02223: [patch] a swap is a swap, but |
ASTERISK-02224: [patch] Using a stream for MusicOnHold |
ASTERISK-02225: Missing statement of licence for included FPM mp3 files |
ASTERISK-02226: [post 1.0] REFER transfer fails with certain hardware (REFER requires NOTIFY) |
ASTERISK-02227: [patch] Problems with () characters in expressions in extensions.conf |
ASTERISK-02228: A @ character as a username for a sip host is read wrong |
ASTERISK-02229: [chan modem] Asterisk internal DTMF detection crashes |
ASTERISK-02230: pbx*CLI> help dial is incomplete |
ASTERISK-02231: [patch] iax2 ignores port and serverport values in iax.conf, iaxprov.conf |
ASTERISK-02232: chan_zap allows you to bridge the two subchannels of a single master |
ASTERISK-02233: sample in configs/extconfig.conf.sample is wrong |
ASTERISK-02234: [patch] Empty messages should not result in a voicemail |
ASTERISK-02235: Mp3Player (for asterisk) doesn't support http:// shoutcast streams |
ASTERISK-02236: [patch] Reboot Grandstreams phones from CLI (proprietary NOTIFY) |
ASTERISK-02237: [patch] Set/show MWI settings for Zap channels from CLI |
ASTERISK-02238: SIGFPE causes asterisk to crash (i4l) |
ASTERISK-02239: Does the new Bugs home work? |
ASTERISK-02240: test 2 |
ASTERISK-02241: [request] make table name for "cdr" table configurable in cdr_odbc |
ASTERISK-02242: res_config_odbc doesn't load full context if id aren't continous (extensions.conf) |
ASTERISK-02243: agi stream file randomly exits with: "ast_waitstream_full: Wait failed (No such file or directory)" message. |
ASTERISK-02244: Patch to properly build and run on NetBSD |
ASTERISK-02245: [patch] h323-sip-mgcp native bridging |
ASTERISK-02246: AGI command "stream file" produces Error on Fedora Core 2 |
ASTERISK-02247: [patch] Small memory leak on unregister of applications |
ASTERISK-02248: [patch] Allow functions to be set |
ASTERISK-02249: [patch] save dialplan does not store CID matching or switch commands |
ASTERISK-02250: Latest cvs causes asterisk to crash |
ASTERISK-02251: [patch] TDM400P FXO: Unexpected control subclass '5' |
ASTERISK-02252: Intel modem 536EP problem |
ASTERISK-02253: [probably not really a bug] IAX2 DTMF not recognized - CVS-HEAD-08/01/04-22:51:56 |
ASTERISK-02254: 1st second of RTP (especially indications) to sip channel is poor quality |
ASTERISK-02255: Makefile has hardcoded paths, should always use paths in /etc/asterisk/asterisk.conf |
ASTERISK-02256: rtptimeout and canreinvite=yes |
ASTERISK-02257: [request] Port Restart |
ASTERISK-02258: typo in astcc |
ASTERISK-02259: extensions reload does not recognize timeout in extensions.conf by save dialplan |
ASTERISK-02260: Turn Callwaiting off on IAXy boxes |
ASTERISK-02261: [patch] show applications like <text> |
ASTERISK-02262: Call parking does not return to original caller after timeout. |
ASTERISK-02263: [patch] Unable to call X-lite using SpeeX |
ASTERISK-02264: zap show channel : provide current call-duration |
ASTERISK-02265: asterisk + mod_php like in Apache |
ASTERISK-02266: [request] Enable option to use ICC compiler instread of GCC. |
ASTERISK-02267: [patch] vmail.cgi modified to work with both voicemail.conf and MySQL voicemail configuration. |
ASTERISK-02268: AGI get_variable doesn't work. |
ASTERISK-02269: [patch] Destorying target span of DACS crashes the kernel |
ASTERISK-02270: Digium FAQ page contains deprecated BYEXTENSION syntax |
ASTERISK-02271: [patch] Add quick login to voicemailmain when user does not enter login |
ASTERISK-02272: [patch] When playing a message longer than X minutes, say the duration in minutes |
ASTERISK-02273: chan_sip does not pass digits while using nat |
ASTERISK-02274: [Request] Please add zap timing method to rtp.c |
ASTERISK-02275: [Patch] Add Confirm Answer (like in chan_zap) into app_dial |
ASTERISK-02276: [patch] Rename events omit Uniqueid |
ASTERISK-02277: app_sms: 1 char missing in directory name. |
ASTERISK-02278: Uniden phones will not work behind a nat |
ASTERISK-02279: callerid field in sip.conf changes Contact: header |
ASTERISK-02280: [request] Allow Asterisk to send the Caller Name in the q931 Facility Message |
ASTERISK-02281: I4L loops on a call that is hung up |
ASTERISK-02282: channel.c 1.134 breaks MOH for MeetMe |
ASTERISK-02283: [patch] chan_iax2.c Flexibility in MYSQL access |
ASTERISK-02284: [patch] MeetMe option to place callers in MOH instead of Hangup when marked user (temporarily) leaves |
ASTERISK-02285: nat=yes saves private ip after new nat=route patch |
ASTERISK-02286: Problem in musiconhold.h while using C++ |
ASTERISK-02287: chan_sip.c: 7731 sip_poke_noanswer: Peer is now UNREACHABLE |
ASTERISK-02288: Inbound SIP call Status |
ASTERISK-02289: [request] Makefiles do not correctly recognize existence of UTRASPARC Family of CPU's. |
ASTERISK-02290: Can only insert one CDR Record... |
ASTERISK-02291: The bugtracker needs a 'forgotten password' feature |
ASTERISK-02292: app_directory incorrectly refers to 'context' when 'dialcontext' has been defined |
ASTERISK-02293: Division by zero error - SIGFPE crash |
ASTERISK-02294: New Error Messages (Also Left Calls) |
ASTERISK-02295: [Patch] Busy/Congestion apps don't work with PRI |
ASTERISK-02296: Error when legacy PBX gets line from te405p |
ASTERISK-02297: no EOF after AGI answer to command: "channel status" |
ASTERISK-02298: Add a courtesy beep to indicate connection to party calling into a parked call |
ASTERISK-02299: [PATCH]: Minor sparc optimiation in build system for libpri |
ASTERISK-02300: [patch] Poke peer when we have IP |
ASTERISK-02301: #define _THREAD_SAFE in localtime.c |
ASTERISK-02302: [patch] Improved ultrasparc support for Asterisk |
ASTERISK-02303: app_festival creates files with permissions set to 000 |
ASTERISK-02304: iaxys still losing registration |
ASTERISK-02305: res_config_odbc, all selects fail |
ASTERISK-02306: Exceptionally long queue length warnings being continuously repeated |
ASTERISK-02307: [request] Put Span Out Of Service from CLI |
ASTERISK-02308: [patch] Add verbosity to pbx.c when it can't find a target triplet in the dialplan |
ASTERISK-02309: [patch] add verbosity to parking timeout event in res_features.c |
ASTERISK-02310: [patch] postgres music on hold |
ASTERISK-02311: Native bridging does Link/Unlink/Link |
ASTERISK-02312: [patch] Bug in handle_add_queue_member |
ASTERISK-02313: [request] immediate=yes / threeway call |
ASTERISK-02314: [request] print to stderr in AGI should also show up in CLI when connecting via "asterisk -r" |
ASTERISK-02315: [Patch] Add Support for the Local channel to astcc |
ASTERISK-02316: chan_zap fails to build under uclibc |
ASTERISK-02317: [patch] Makefile cleanup a bit... |
ASTERISK-02318: Use count doesn't decrease in format_wav_gsm.c |
ASTERISK-02319: [patch] add verbosity and a warning to app_voicemail |
ASTERISK-02320: app_read.c outputs garbage in console |
ASTERISK-02321: [patch] Add option to force immediate password change if user has a specific password |
ASTERISK-02322: Members count of calls taken is wrong when *8 is used |
ASTERISK-02323: [not an asterisk bug] Malformed 401 Message from SER with bindaddr=0.0.0.0 and asterisk coneccted to two local networks |
ASTERISK-02324: [request] Add sound file playback to Dial Application option c |
ASTERISK-02325: [patch] add externpass cmd and voicemail reload |
ASTERISK-02326: [request] [post 1.2] support reponses to INVITEs on IP address aliases to come back on the same IP |
ASTERISK-02327: TDM400P modules do not recognize lack of cable/dial tone |
ASTERISK-02328: [patch] Add temporary greetings to voicemail |
ASTERISK-02329: Please make TON (Type of Number) accessible in asterisk |
ASTERISK-02330: Changes to chan_zap.c version 1.331 break channel bank FXO ports |
ASTERISK-02331: [Patch] exitcontext option does not work if 'o' or 'a' not in current context |
ASTERISK-02332: [patch] Make the codecs we answer with configurable |
ASTERISK-02333: [patch] output time in hms a channel has been active when cli 'show channels' is called. |
ASTERISK-02334: [patch] Add mysqlcanblock= and mysqlsipfriends= to chan_sip |
ASTERISK-02335: missing symbol in latest cvs |
ASTERISK-02336: Trouble with configuration files when entering ODBC connection strings |
ASTERISK-02337: SIP 400 response "Missing/Invalid From" -> DIALSTATUS="NOANSER" |
ASTERISK-02338: IAXY generates incorrect timestamps in IAX2 stream |
ASTERISK-02339: no out bound audio not even on echo test |
ASTERISK-02340: [patch] append hostname to logfiles |
ASTERISK-02341: [patch] couple of voicemail password change bugs |
ASTERISK-02342: g.729 show license is not working |
ASTERISK-02343: [Feature + Patch] E-mail info and list for astcc |
ASTERISK-02344: [Patch] Wait until # is press before making the call go throught |
ASTERISK-02345: [patch] interrupt user and admin menu by choosing option |
ASTERISK-02346: [Patch] Allow the use of L(x) flag in Background to play a specified language |
ASTERISK-02347: [patch] Zap attached devices no long receive callerid |
ASTERISK-02348: [patch] -t flag to asterisk args by anthm |
ASTERISK-02349: [patch] add format_mp3, format_slinear and format_base64_wav_gsm by anthm |
ASTERISK-02350: [Patch] Make w flag work better, and say prompt on enter/leave of marked |
ASTERISK-02351: [patch] res_monitor patch by anthm |
ASTERISK-02352: [post-1.0] [patch] res_sqlite, adds sqlite_switch, cdr engine, cli tools and SQL dialplan application by anthm |
ASTERISK-02353: [patch] if no password set, don't bother asking |
ASTERISK-02354: [patch] prevent a user changing mailbox password |
ASTERISK-02355: [Patch] Switch from user to admin in meetme |
ASTERISK-02356: alloca.h in utils |
ASTERISK-02357: usedistinctiveringdetection in X100P analog cards |
ASTERISK-02358: Compile failure in say.c since cvs version 1.35 |
ASTERISK-02359: SIP response 503 "Service Unavailable" -> DIALSTATUS = NOANSWER |
ASTERISK-02360: Avoided deadlock for Zap channel |
ASTERISK-02361: [general] General Holding bug for BSD/OSX Compatibility |
ASTERISK-02362: Problem with DTMF being passed from Cisco GW to asterisk on ingress calls |
ASTERISK-02363: [patch] callerid.c TELEKOM callerid |
ASTERISK-02364: [patch] Modify level of log entry related to GotoIf when no branch taken |
ASTERISK-02365: [patch] fix padding of field in iax2 debug output to aid readability |
ASTERISK-02366: [request] [patch] Lightweight ODBC API for asterisk |
ASTERISK-02367: When dialling #, asterisk gets %23 |
ASTERISK-02368: Call progress detection for Costa Rica is broken |
ASTERISK-02369: rtptimeout hanging up "dialling channels" |
ASTERISK-02370: One-way audio when transferring calls after they've been picked up by a queue |
ASTERISK-02371: [PATCH TRIVIAL] Make action_getvar() respect ActionID. |
ASTERISK-02372: [PATCH TRIVIAL] Make action_getvar() respect ActionID. |
ASTERISK-02373: Asterisk Manager response corruption. |
ASTERISK-02374: dynamic and default queue members call counters respond different to a reload. |
ASTERISK-02375: Asterisk Deadlock |
ASTERISK-02376: geting callingpres status |
ASTERISK-02377: Macro support in Dial() |
ASTERISK-02378: [patch] SIP Headers - Modified Format Handler |
ASTERISK-02379: ODBC() application for asterisk. |
ASTERISK-02380: Incorrect Destination written in CDR when Macro() used with Goto()command |
ASTERISK-02381: ast_channel_walk_locked avoiding a deadlock causes breaks in audio |
ASTERISK-02382: GSM sounds fail to play on GSM call |
ASTERISK-02383: When leaving a VM, 'accepting' recording doesn't hang up on person. |
ASTERISK-02384: app_mp3 used in MP3Player fails to play stream due to timeout |
ASTERISK-02385: * dont know ipaddress of dynamic sip peers after restart |
ASTERISK-02386: DIALSTATUS through IAX trunk? |
ASTERISK-02387: Bug introduced in CVS with the guardtime feature |
ASTERISK-02388: problem with sip registration (to sipgate.de) with pedantic=yes |
ASTERISK-02389: [PATCH] Automate import of trunks |
ASTERISK-02390: calls and audio recorded from GS phone plays back at double speed |
ASTERISK-02391: Who would like agents to be able to respond to emails as well as calls? |
ASTERISK-02392: [post-1.0][patch]Move 'lastcallerid' into 'struct ast_channel' |
ASTERISK-02393: call-id header with FQDN or externip |
ASTERISK-02394: [patch] Add 'set debug n' to cli and tweak 'set verbose n' |
ASTERISK-02395: [post 1.0] [patch] ODBC voicemail, support of HTML messages |
ASTERISK-02396: iax2 show channels report too great lag |
ASTERISK-02397: NOTICEs from sched.c |
ASTERISK-02398: ringing tones not transmitted |
ASTERISK-02399: regexten feature improvement request |
ASTERISK-02400: [patch] make manager originate send status in case of asynchronous execution. |
ASTERISK-02401: Matching on dialstatus in the dialplan as laid out in extensions.conf.sample does not function. |
ASTERISK-02402: voicemail file sequencing problem |
ASTERISK-02403: Add possibility to enable/disable sip friends from sip.conf |
ASTERISK-02404: Missing simicolon in latest chan_sip (CVS rev 1.502) |
ASTERISK-02405: MYSQL Voice Mail Lookup |
ASTERISK-02406: Karma changes (please ignore) |
ASTERISK-02407: trunk not always used |
ASTERISK-02408: [request] auto limit IAX2 trunked calls or autoincrease the value to avoid flooding warning messages. |
ASTERISK-02409: after latest CVS update, MP3Player command giving RTP Bad Packet Errors |
ASTERISK-02410: compatability problem with huawei sip interface. |
ASTERISK-02411: [patch] channel.c 'timeleft' uses non-existant sound files |
ASTERISK-02412: SIP and h323 can't handle LPC10 |
ASTERISK-02413: Call rejected -> DIALSTATUS=NOANSWER |
ASTERISK-02414: RTP is not immediately transmitted after 183 session progress is sent |
ASTERISK-02415: [post-1.0 patch] ANI II exposed as a VAR in dialplan. |
ASTERISK-02416: app_disa no longer restores dialtone after ignorepat |
ASTERISK-02417: Parking calls with blind transfer phones like GrandStream does not read digits |
ASTERISK-02418: Including a 'w' in the dial string on a PRI trunk breaks calling number in Q.931 frame? |
ASTERISK-02419: [patch] Pick-up extension is not configurable |
ASTERISK-02420: [patch] strncpy with wrong sizeof in config parsing |
ASTERISK-02421: "make update" don't use compression |
ASTERISK-02422: Answer() can interfere with E&M Wink signalling on cT1 |
ASTERISK-02423: Asterisk does not stimulate a far end disconnect when Hangup() called on E&M trunk during handshake |
ASTERISK-02424: [PATCH] Client application to check Usage Info |
ASTERISK-02425: [patch] Attended Pound Transfers The Perfect New Years Resolution!!! |
ASTERISK-02426: Authentication _sometimes_ fail with clients |
ASTERISK-02427: Suggested clarification in app.h |
ASTERISK-02428: [PATCH] Add advance call timeout warnings |
ASTERISK-02429: [PATCH] Reset inuse info in card database |
ASTERISK-02430: [patch] to call cli functions from agi script |
ASTERISK-02431: [new app] WaitForSilence application |
ASTERISK-02432: Segmentation fault in chan_iax2.c |
ASTERISK-02433: MYSQL Voice Mail Lookup and IAX Friends |
ASTERISK-02434: Unable to pass parameters of macro into Dial (from AGI script). |
ASTERISK-02435: [patch] initial Remote-Party-ID support & app to set RestrictCID |
ASTERISK-02436: [patch] Voicemail login causes change to invalid/corrupt language code |
ASTERISK-02437: Manager Redirect action does not function in certain conditions. |
ASTERISK-02438: [patch] Allow app_directory to work with REALTIME |
ASTERISK-02439: [patch] vmail.cgi does not use MySQL database when enabled. |
ASTERISK-02440: ZT_TIMERPING |
ASTERISK-02441: Linking wrong threads library and other threads issues |
ASTERISK-02442: With rpid "privacy=full" callerid is always "Unknown" <unknown> |
ASTERISK-02443: [patch] term.c - support Eterm colors |
ASTERISK-02444: No response -> NO ANSWER |
ASTERISK-02445: the CALLERIDNAME information not populated on a PRI Call |
ASTERISK-02446: asterisk SIP ignores SIP packets from peer in "early" state |
ASTERISK-02447: Patch 0002434 can cause broken event output. |
ASTERISK-02448: waw49 files sometimes get size for data chunk on record |
ASTERISK-02449: App Sox (new app -anthm) |
ASTERISK-02450: Data through TE410P stops for few seconds |
ASTERISK-02451: [patch] Caller ID wrong on call transfer |
ASTERISK-02452: res/res_agi.c doesn't compile |
ASTERISK-02453: [patch] Eject last user |
ASTERISK-02454: change in chan_mgcp.c causes dtmfmode=inband to not work |
ASTERISK-02455: Update Mantis |
ASTERISK-02456: [patch] Detect terminals which support color |
ASTERISK-02457: rtp.c generates seemingly erroneous warning message |
ASTERISK-02458: SIP NOTIFY returned during Supervised Transfer not according to RFC |
ASTERISK-02459: [patch] AGI Debugging support |
ASTERISK-02460: rtp.c warning - rtp structure is null |
ASTERISK-02461: [patch] Holdtime incorrectly announced as '1 minutes' when holdtime less than 2 minutes |
ASTERISK-02462: [patch] Fix 'show applications like <string>' |
ASTERISK-02463: [discuss] Race condition in timed includes |
ASTERISK-02464: [patch] Queue is not ejecting callers if all members are logged off. |
ASTERISK-02465: An unavailable CODEC is sometimes accepted |
ASTERISK-02466: Asterisk Manager Proxy -- simpleproxy.pl |
ASTERISK-02467: [patch] "Thank you" message should not be played after first position announcement |
ASTERISK-02468: app_saycount.pl.c - an application that returns the right counting word for a number in Polish |
ASTERISK-02469: [patch] cdr.txt typo |
ASTERISK-02470: [patch] Override position announcement when there is nobody waiting in the queue |
ASTERISK-02471: Transferring Calls |
ASTERISK-02472: [patch against 1.0] res_agi needs socket defs |
ASTERISK-02473: compilation against latest ilbc don't work |
ASTERISK-02474: [patch] cleaned up cdr_mysql.c |
ASTERISK-02475: No voice on AT&T 4ess trunkgroup |
ASTERISK-02476: Non d-channel T's not working on AT&T trunk |
ASTERISK-02477: Provide configurable timers for PRI's |
ASTERISK-02478: Ye old: 'Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)' |
ASTERISK-02479: [patch] Send email messages from within asterisk |
ASTERISK-02480: [request] SNOM 200 CMC code to UserData field in CDR |
ASTERISK-02481: reload and extensions reload forget subscriptions for presence (e.g. SNOM busy lamps) |
ASTERISK-02482: [request] Record calls at ZAP interface |
ASTERISK-02483: Missing Curly braces appear to cause sip_reg_timeout problem (numerous SIP channels left open) |
ASTERISK-02484: id3 version 2 tags on Free Play Music files crash asterisk 1.0 (and CVS) on gentoo |
ASTERISK-02485: Asterisk does not respond to BYE message when context is set to canreinvite=Yes |
ASTERISK-02486: [request] priority of same and next in extensions.conf |
ASTERISK-02487: [request] Can MYSQL_LOGUNIQUEID be automatic... |
ASTERISK-02488: Code cleanup, extensions including regex matching |
ASTERISK-02489: [patch] Add channel group setting to Dial() |
ASTERISK-02490: res_agi.c will not compile on FreeBSD |
ASTERISK-02491: [patch] new jitter buffer |
ASTERISK-02492: [patch] app_callback - calls users back. |
ASTERISK-02493: app_math adds Sum, Multiply, Divide, Subtract, Modulus, GT, LT, GTE, LTE, EQ functions to asterisk |
ASTERISK-02494: 2342 bug fix breaks chan_zap answer behavior |
ASTERISK-02495: [patch] Adjustabe Speex Codec |
ASTERISK-02496: Linux 2.6: Timing off when using USB devices |
ASTERISK-02497: AGI network scripts (agi:// URLs) do not parse ports correctly |
ASTERISK-02498: AGI network scripts (agi:// URLs) do not parse ports correctly |
ASTERISK-02499: type-punned pointer will break strict-aliasing rules ... hash/ndbm.c out of date ? |
ASTERISK-02500: [patch] remove warning from localtime.c |
ASTERISK-02501: Design flaw in chan_sip |
ASTERISK-02502: recent "less than" changes broken |
ASTERISK-02503: [patch] Fix endless loop in card generation when starting digit=0 |
ASTERISK-02504: [Patch] Undocumented feature now documented |
ASTERISK-02505: [patch] French update for say.c |
ASTERISK-02506: [request] chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 |
ASTERISK-02507: Ignore ;transport=udp from 302 Moved Temp. requests |
ASTERISK-02508: [patch] IAX2 Native Bridge Crashes |
ASTERISK-02509: No sound generated by asterisk (prompts) |
ASTERISK-02510: Ignore message type 6 in CID delivery |
ASTERISK-02511: SIP Early Media Not Working |
ASTERISK-02512: [PATCH] Assorted additions - SIP&IAX Info |
ASTERISK-02513: silence when logged in (second) via SIP from Grandstream HT 286 into Meetme |
ASTERISK-02514: Transfer blocks new agent calls |
ASTERISK-02515: app_vareval - allows the evaluation of dynamically built variables |
ASTERISK-02516: STREAM FILE and GET DATA not work in current cvs |
ASTERISK-02517: [External Module] res_sqlite for SQLite 3 |
ASTERISK-02518: Has(New)VoiceMail does not assume INBOX if no folder specified |
ASTERISK-02519: NetBSD build doesn't always find libncurses |
ASTERISK-02520: [patch] Rework ast_app_has_voicemail to handle more cases |
ASTERISK-02521: extens with cidmatch no longer work on zap |
ASTERISK-02522: [patch] Delay member connect to caller |
ASTERISK-02523: [patch] Report caller's hold time to agent |
ASTERISK-02524: [patch] MeetMe needs configurable Music On Hold |
ASTERISK-02525: SIP_CODEC variable not working in 1.0.0 and 1.0.1 |
ASTERISK-02526: PRES_NUMBER_NOT_AVAILABLE is not defined in pri_pres2str |
ASTERISK-02527: [patch] chan_sip.c will not properly compile with latest callerid changes and MYSQL support |
ASTERISK-02528: [patch] ast_true in cdr_odbc |
ASTERISK-02529: [patch]RTP debugging |
ASTERISK-02530: chan_mgcp crashes * CVS-HEAD-09/29/04 |
ASTERISK-02531: Asterisk doesn't build with uclibc |
ASTERISK-02532: app_queue incorrectly reports that no one is answering the queue |
ASTERISK-02533: Asterisk will dergister all SIP phones but will respond to cli cmds |
ASTERISK-02534: Latest CVS (2004-10-05 12:30) crashes if an attempt is made to transfer a call. |
ASTERISK-02535: [patch] mysql log of queue_log |
ASTERISK-02536: Allow "c" dial flag to work with non-zap channels |
ASTERISK-02537: chan_mgcp binds rtp to 0.0.0.0 insted of addr, specified in "bindaddr" option |
ASTERISK-02538: notification msg during a blind transfer does not include the registrar's IP address |
ASTERISK-02539: Discontiguous descending Zap channel group fails to use all members |
ASTERISK-02540: SIP MWI not working with Xten eyeBeam and possibly other clients |
ASTERISK-02541: Audio from PBX not heard on Polycom IP500/SIP phone with current release or -HEAD |
ASTERISK-02542: [poet 1.0] [new app] app_realtime |
ASTERISK-02543: TRON TROFF in extensions.conf |
ASTERISK-02544: [patch] Always fflush for any CDR file (not only for Master.csv) |
ASTERISK-02545: garbled audio with trunking on some devices |
ASTERISK-02546: Typos on Download page |
ASTERISK-02547: RDNIS is always empty |
ASTERISK-02548: chan_sip 1.521 breaks SIP NOTIFY |
ASTERISK-02549: receiving gethostbyname() error on call to festival in dialplan |
ASTERISK-02550: [patch] Count for show modules |
ASTERISK-02551: [patch] voicemail beep shouldn't be played till we get next message number |
ASTERISK-02552: [patch] memset fixes |
ASTERISK-02553: Problem with MWI in the bugfix of chan_sip.v ver 1.520 - 1.521 |
ASTERISK-02554: [patch] app_sms doesn't use paths from asterisk.conf |
ASTERISK-02555: attribution missing in CHANGES |
ASTERISK-02556: [patch] Memory leak fixes for chan_sip.c with realtime() functions |
ASTERISK-02557: ILBC produces choppy sound |
ASTERISK-02558: [patch] Add mute & umute ALL |
ASTERISK-02559: [patch] Fix two trivial verbosity related issues |
ASTERISK-02560: [patch] Elaborate on where rejected IAX2 call was trying to reach |
ASTERISK-02561: No Audio / Fast busy on incoming call on T100P |
ASTERISK-02562: Nortel SIP dtmf |
ASTERISK-02563: Developer documentation for realtime config |
ASTERISK-02564: [devel branch] asterisk segfault on reload |
ASTERISK-02565: cidmatch broken when pattern contains ranges |
ASTERISK-02566: [patch] app_lookupcidname borken (CVS rev 1.4 of app_lookupcidname.c) |
ASTERISK-02567: [patch] chan_iax2 shouldn't use IAX/Registry on temponly peers |
ASTERISK-02568: Can't use virtual IP's for VOIP |
ASTERISK-02569: [request] Detect dialtone on Zap channels before dialing |
ASTERISK-02570: res_config_mysql - realtime driver |
ASTERISK-02571: h323 channel make error |
ASTERISK-02572: Patch to modify queue message system |
ASTERISK-02573: chan_sip should reply FROM the same address that request was sent TO |
ASTERISK-02574: [patch] set verbose and set debug still conflict with each other |
ASTERISK-02575: [patch] don't seed p->temponly sipfriends when using realtime |
ASTERISK-02576: [patch] Remove VoiceMail2 and VoiceMailMain2 backwards compat in cvs-head |
ASTERISK-02577: SIP INVITE header doesn't include number to dial. |
ASTERISK-02578: qualify != 'no' in realtime (sipfriends extconfig) make the peer unreachable and sip show peer PEER_NAME endup hanging Asterisk |
ASTERISK-02579: All Dial commands fail after OSPLookup |
ASTERISK-02580: [v1-0] Recent callToken changes break send_digit |
ASTERISK-02581: Cisco BTS problem |
ASTERISK-02582: [patch] app_realtime - add CLI debug abilities |
ASTERISK-02583: [patch] terminator key for app_record |
ASTERISK-02584: [patch] Add channel variable to BackgroundDetect / app_talkdetect |
ASTERISK-02585: AGI Application tends to become non-responsive |
ASTERISK-02586: chan_h323 doesn't initiate calls at all |
ASTERISK-02587: [request] "no call progress" indication timeout option. |
ASTERISK-02588: [patch] Fixes to support attended transfers from a libiax client |
ASTERISK-02589: Record application fails with ast_set_read_format: Unable to find a path from GSM to UNKN |
ASTERISK-02590: tab completion on show dialplan from -r asterisk with large extensions.conf will segfault |
ASTERISK-02591: [patch] more details for sip show peers |
ASTERISK-02592: [patch] preliminary zap reload from cli (no signalling, just parameters) |
ASTERISK-02593: [patch] Possibility to disable some parts of the announce |
ASTERISK-02594: [patch] Native MOH without mpg123 |
ASTERISK-02595: usage count isn't decremented when agent/queue call completed |
ASTERISK-02596: SetLanguage without parameter segfault |
ASTERISK-02597: Segfault when res_config_odbc and res_config_mysql are load |
ASTERISK-02598: [patch] show module like <keyword> |
ASTERISK-02599: [patch] Improved debug output in SIP and IAX (lagging, RSA key info) |
ASTERISK-02600: [PATCH] Tweak 'pri debug' so output is displayed even when verbosity = 0 |
ASTERISK-02601: Directory Application doesn't work if using realtime config. |
ASTERISK-02602: Voicemail configuration still pulls zone info from voicemail.conf if using realtime config. |
ASTERISK-02603: Voicemail notification is not sent to e-mail/pager if using realtime config. |
ASTERISK-02604: BSD systems fail with "bindaddr=0.0.0.0" |
ASTERISK-02605: Dialing from keypad in gnomemeeting crashes asterisk |
ASTERISK-02606: [request] presence/hint for queues |
ASTERISK-02607: chan_h323 core dumps |
ASTERISK-02608: GotoIfTime does not support new "n" priority |
ASTERISK-02609: [patch] update to readme |
ASTERISK-02610: [patch] Makefile tweak to point to kernel build in /lib/modules/VERSION/build |
ASTERISK-02611: c option is not documented in show application dial |
ASTERISK-02612: [patch] Update of README |
ASTERISK-02613: Breakage of ChanIsAvail because of bad hangup handling... |
ASTERISK-02614: Chan_H323 dials out incorrectly |
ASTERISK-02615: Dial(Zap/1) without timeout gives dialtone |
ASTERISK-02616: [patch] app_festival calls ast_destroy on cfg before it's done |
ASTERISK-02617: sip stops communicating when gethostbyname() temporarily fails |
ASTERISK-02618: tor2.c tor2_probe |
ASTERISK-02619: [patch] fix oops on reload chan_zap.so if we're using pseudo channels. |
ASTERISK-02620: [PATCH] Make operator=no feature of app_voicemail behave as documented |
ASTERISK-02621: The version 1.29 of loader.c crash on the apllication app_qcall and makr undefined symbol |
ASTERISK-02622: [isdn4linux] echotest / voicemail crash when receiving DTMF from the PSTN[ |
ASTERISK-02623: [patch] Use of bindaddr=0.0.0.0 broken under BSD |
ASTERISK-02624: After #2650 fix chan_h323 doesn't build anymore :( |
ASTERISK-02625: 487 Message not sent after receipt of CANCEL |
ASTERISK-02626: Fax redirect does not always work |
ASTERISK-02627: Monitor with option 'b' records silence when bridging Zap channels |
ASTERISK-02628: CVS HEAD fails compile. Dundi problem? |
ASTERISK-02629: [patch] wrong logging statement when sending voicemail alert email |
ASTERISK-02630: help for http://bugs.digium.com/bug_report_page.php 403's |
ASTERISK-02631: [patch] app_forkcdr output beautifications |
ASTERISK-02632: [patch] ShowVars() function call OR try app_dumpchan.c |
ASTERISK-02633: [patch] libpri compiles with wrong byte endian on FreeBSD |
ASTERISK-02634: pbx_dundi.c doesn't compile out of the box on NetBSD |
ASTERISK-02635: meetmeadmin/admin_exec() references null command pointer |
ASTERISK-02636: 'asterisk -r -x "show channels"' often returns no channel information |
ASTERISK-02637: [patch] SayUnixTime and SayNumber to say in British and Norwegian syntax |
ASTERISK-02638: Fresh asterisk cvs won't compile - suspect lock.h |
ASTERISK-02639: app_queue rings dynamic agents who are already on the phone |
ASTERISK-02640: [PATCH] vmail.cgi patch to work with newer realtime config db name, and fixes taint problems when forwarding voicemails |
ASTERISK-02641: [patch] ACK sent to wrong address |
ASTERISK-02642: Country tones for Singapore |
ASTERISK-02643: Country tones for Singapore |
ASTERISK-02644: res_perl does not compile with CVS |
ASTERISK-02645: Add preprocessor parsable ASTERISK_VERSION |
ASTERISK-02646: [new_app] app_intercept.c |
ASTERISK-02647: show channels crashes asterisk |
ASTERISK-02648: [reqest] backgrounddetect also could listen for busy tones |
ASTERISK-02649: Progress indicator must be optional in 'PROGRESS' |
ASTERISK-02650: Crash in ast_queue_frame (channel.c) |
ASTERISK-02651: dialog matching problems |
ASTERISK-02652: asterisk "101 Dialog Establishement" error |
ASTERISK-02653: rtp.c error - unknown RTP codec 127 received |
ASTERISK-02654: Agents taking Calls |
ASTERISK-02655: [patch] Add manager cmd - Action: Agents |
ASTERISK-02656: port of pbx/pbx/dundi.c reset_global_eid() to FreeBSD |
ASTERISK-02657: Asterisk Manager API (sockets) |
ASTERISK-02658: ast_modem_pvt() uses memory before checking if it is valid |
ASTERISK-02659: callfile, etc: Jump into extension without needing to have already established call |
ASTERISK-02660: Add Israeli tone zone |
ASTERISK-02661: request for dialing feature 'exten => (555)1234,1,app' |
ASTERISK-02662: Sending SMS with no hangup supervision will hang the channel |
ASTERISK-02663: False file-not-found error in voicemail |
ASTERISK-02664: [patch] Chan Zap close bug |
ASTERISK-02665: ZapRas/HDLCPPP is broken |
ASTERISK-02666: RE bug # 0002688, New Country Indications for Singapore are likely wrong |
ASTERISK-02667: [patch] Application UserEvent does not populate the 'body' parameter |
ASTERISK-02668: [Patch] add a flag to app_dial to wait until someone has answer and confirm |
ASTERISK-02669: Asterisk Makefile -DDEBUG_THREADS generating vast number of warnings. |
ASTERISK-02670: Use the newly added FreeBSD MAC-address lookup for NetBSD |
ASTERISK-02671: AudioCodes with last firmware couldnt register on Asterisk |
ASTERISK-02672: joinempty does not work with members defined statically |
ASTERISK-02673: [patch] Stack applications |
ASTERISK-02674: Calls hung up when agent parks... |
ASTERISK-02675: [patch] N+101 on failed playback |
ASTERISK-02676: [patch] fix app_voicemail when using wav49 |
ASTERISK-02677: asterisk says telephone-event when apps want to hear PCMU. |
ASTERISK-02678: Logging line in chan_local needs variables flipped |
ASTERISK-02679: Asterisk sending PRI restarts every hour |
ASTERISK-02680: latest v1-0 CVS chan_h323 doesn't want to compile on RH9 |
ASTERISK-02681: Parking a call crash the asterisk |
ASTERISK-02682: [patch] Add distribution lists to app_voicemail |
ASTERISK-02683: h323 calls gets recorded by voicemail application at double speed if voicemail format is .wav |
ASTERISK-02684: comma "," gets changed to pipe "|" |
ASTERISK-02685: [request] new cmd IF |
ASTERISK-02686: Problem with multiple lines on Cisco phones. |
ASTERISK-02687: Add pkg-config support to asterisk |
ASTERISK-02688: No call pickup possible... |
ASTERISK-02689: [patch] CLI command 'pri restart span <spannum>' |
ASTERISK-02690: [patch]Allow AGI script to get HANGUPCAUSE variable |
ASTERISK-02691: Asterisk hangs from time to time |
ASTERISK-02692: cli summaries for h.323 show tokens and h.323 hangup swapped |
ASTERISK-02693: [patch] Patch to make Asterisk work on Solaris |
ASTERISK-02694: [patch] Chan SIP Transfer CRASH |
ASTERISK-02695: zaptel/README is missing a prerequisite |
ASTERISK-02696: [PATCH] DUNDi support for 1.0.2 release |
ASTERISK-02697: [patch] zaptel/zaptel.conf.sample doesn't specify all zones |
ASTERISK-02698: [patch] logger.c casts pthread_self() to long |
ASTERISK-02699: nonce generation errors |
ASTERISK-02700: SIP hold/transfer fails |
ASTERISK-02701: [patch] Give warning on compile with old libpri and new QSIG |
ASTERISK-02702: [patch] Say number for portuguese is wrong |
ASTERISK-02703: [request] when dial connects, fork/reset the CDR. |
ASTERISK-02704: Zaptel documentation doesn't say anything on how to set the card to an E1 |
ASTERISK-02705: No queue announcements when MOH is replaced by ringing |
ASTERISK-02706: Festival not working for Asterisk 1.0 |
ASTERISK-02707: Wrong Request URI - RFC3261 |
ASTERISK-02708: [patch] make table name for "cdr" table configurable in cdr_odbc and cdr_odbc module unloadable |
ASTERISK-02709: Incoming calls handling, FastStart & H245 state |
ASTERISK-02710: [patch] Add depend support to h323/ast_h323 |
ASTERISK-02711: [patch] libpri debugging cleanup |
ASTERISK-02712: [patch] PRI progress indicator support |
ASTERISK-02713: [patch] PRI redirecting number IE support |
ASTERISK-02714: app_record is broken saying that filename argument is not present |
ASTERISK-02715: [patch] Little code cleanups for current CVS-HEAD |
ASTERISK-02716: [patch] Passing REDIRECTING NUMBER IE on PRI outgoing calls |
ASTERISK-02717: [patch] Don't show partially-filled strings over remote console (asterisk -r) |
ASTERISK-02718: [patch] More output for debugging purposes |
ASTERISK-02719: Patch to app_disa for stutter-dialtone and response / digittimeout. |
ASTERISK-02720: presence/hint as an application |
ASTERISK-02721: Erroneous warning Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout) |
ASTERISK-02722: after asterisk upgrade it reports error and doesn't load anymore. |
ASTERISK-02723: getting a (matching?) query while dundi debug is on crashes asterisk |
ASTERISK-02724: Incomming caller*id presentation |
ASTERISK-02725: Asterisk Crash/Core Dump |
ASTERISK-02726: IAX2 trunked channel cannot recover from loss of packet containing 'first full voice frame'? |
ASTERISK-02727: h323 calls gets recorded by voicemail application at double speed if voicemail format is .wav |
ASTERISK-02728: [patch] VMAuthenticate |
ASTERISK-02729: [patch] STREAM FILE supports a timeout in res_agi.c |
ASTERISK-02730: Make channel-vars readable with pbx_builtin_getvar_helper |
ASTERISK-02731: Trouble with music on hold when fname_base contains special characters |
ASTERISK-02732: Unknown RTP codec 72 received |
ASTERISK-02733: [patch] German syntax for say.c and it's neighborhood; Creation of say_enumeration (1st,2nd,...,101st,...,last) |
ASTERISK-02734: Syntax error before * token in file included from chan_phone.c:37: |
ASTERISK-02735: Zaptel crashes when unloaded, then reloaded and alarm cleared |
ASTERISK-02736: [PATCH] allow zap gain and echo params to be twiddled on the fly from console |
ASTERISK-02737: [request] automatic disabling echo cancellation when data connection is on a call |
ASTERISK-02738: chan_h323 will not compile |
ASTERISK-02739: [patch] Generic Digits preliminary support |
ASTERISK-02740: [patch] - Fix correct output of sendpage debug |
ASTERISK-02741: compilation error in chan_h323 in 1.0.2 stable version |
ASTERISK-02742: variables are lost in a blind transfer |
ASTERISK-02743: libpri no longer detects remote answer |
ASTERISK-02744: SIP Notify / Message waiting does not conform to RFC. |
ASTERISK-02745: AMIS networking |
ASTERISK-02746: Callier ID Name is lost when calling from SIP channel to Zap channel. |
ASTERISK-02747: [PATCH] added support for pins to astcc. |
ASTERISK-02748: [PATCH] New agi script ASTPP |
ASTERISK-02749: Teach zaptel's Makefile to _actually_ install into arbitary locations |
ASTERISK-02750: [patch] Do not use call progress analysis on PRI links |
ASTERISK-02751: [patch] Frame debugging enhancements |
ASTERISK-02752: oneway audio with iaxy due to CODEC mismatch caused by iax.conf |
ASTERISK-02753: SIP REGISTER ignores 'authuser' setting |
ASTERISK-02754: Asterisk build fails on Darwin/OSX |
ASTERISK-02755: temporarily set SIP UA string on CLI |
ASTERISK-02756: WCUSB driver doesnt work |
ASTERISK-02757: Cisco CID blocking cause crash |
ASTERISK-02758: Change HOSTCC to CC |
ASTERISK-02759: [patch] app behavioral modifications |
ASTERISK-02760: [patch] fix seeding verbose |
ASTERISK-02761: Spelling error |
ASTERISK-02762: Request for decimal rates |
ASTERISK-02763: Request for decimal rates |
ASTERISK-02764: Supervised transfer will result into called party hearing moh |
ASTERISK-02765: [patch] Add timezone support for IAX date/time Information Element |
ASTERISK-02766: config from db via include |
ASTERISK-02767: chan_sip segfaults on incoming call from FWD. |
ASTERISK-02768: Include in extensions.conf seems to be broken with asterisk-1.0.1 and 1.0.2 |
ASTERISK-02769: Voicemail doesn't work with mailbox names beginning with "u" |
ASTERISK-02770: Voicemail doesn't work with mailbox names beginning with "u" |
ASTERISK-02771: [patch] correctly reinit a variable and other more substantial modifications to MARK2 echo canceller |
ASTERISK-02772: Incomplete CDR record when originating call from manager |
ASTERISK-02773: [patch] Support for Note2 Table 4-3/Q.931 (some information elements may be repeated) |
ASTERISK-02774: [patch] clean up a bit in chan_sip |
ASTERISK-02775: [PATCH] app_vmoutcall |
ASTERISK-02776: [patch] Allow globbing in #include on config files |
ASTERISK-02777: [patch] Support for disabling detection of certain call progress tones |
ASTERISK-02778: goertzel dsp was off-by-one |
ASTERISK-02779: outgoing spool directory dials number correctly then need way to dial extension like D(123) from app_dial |
ASTERISK-02780: add option to app_chanisavail for "in use" |
ASTERISK-02781: [patch] expose dsp->tstate and dsp->count if needed. |
ASTERISK-02782: [PATCH] chan_phone.c does not compile with 2.6 kernel headers |
ASTERISK-02783: [PATCH] FD not close after getting default local EID |
ASTERISK-02784: Asterisk 1.02 quits upon launch on Darwin/OSX |
ASTERISK-02785: pedantic=yes makes calls to voip providers impossible |
ASTERISK-02786: [patch] Patterns may include -, but extensions may not |
ASTERISK-02787: [patch] Fullcontact fixes |
ASTERISK-02788: [patch] Make mailbox check time configurable |
ASTERISK-02789: [patch] New app in chan_sip: sipgetheader() |
ASTERISK-02790: expose timing functions from pbx.c for use in other appliactions |
ASTERISK-02791: [patch] Italian date syntax for say.c |
ASTERISK-02792: Blind call transfers on SIP channels don't work from within AGI app |
ASTERISK-02793: [patch] Italian date syntax for app_voicemail.c |
ASTERISK-02794: Faster ADPCM and G726-32 codec code |
ASTERISK-02795: User control of specific authentication methods |
ASTERISK-02796: If voicemail.conf #includes additional files, when you try to change PIN it doesn't work. |
ASTERISK-02797: [patch] SipAddHeader() Application |
ASTERISK-02798: [patch] - Code to allow reverse polarity to indicate a hangup on the channel |
ASTERISK-02799: language=de can't chage the language in sip.conf |
ASTERISK-02800: [patch] allow setting permissions so vmail.cgi can read vmailbox |
ASTERISK-02801: SIP messages contain 0.0.0.0 as IP for Asterisk |
ASTERISK-02802: [patch] Preset Channel Vars In Users/Friends In iax.conf/sip.conf |
ASTERISK-02803: [patch] it allows to send the response 180 Ringing even if has been already sended '183 Progress Response' |
ASTERISK-02804: Type of Number not in CVS available as claimed |
ASTERISK-02805: [patch] fix building on older linux systems |
ASTERISK-02806: [patch] Formatting and additional comments |
ASTERISK-02807: [request] Keeping Asterisk in the Channel Path using DUNDi and notransfer=yes |
ASTERISK-02808: [patch] file.c const const warning |
ASTERISK-02809: [patch] make res_crypto less chatty under valgrind. |
ASTERISK-02810: [patch] Line 227 & 228 in res_musiconhold.c wtf are we closing really? |
ASTERISK-02811: [patch] Outbound proxy support |
ASTERISK-02812: bad dependency on order of contexts in sip.conf |
ASTERISK-02813: [patch] cli.c tweaks |
ASTERISK-02814: Crash when terminating several calls at the same time |
ASTERISK-02815: [branch] RTCP-support |
ASTERISK-02816: [patch] add -e option to exec something after bootup |
ASTERISK-02817: [patch] deadlock H.323 |
ASTERISK-02818: [patch] libiax2 updates: CNG, MSVC, XFER, ALIGN, codecs, misc |
ASTERISK-02819: [patch] leavewhenempty doens't work |
ASTERISK-02820: [patch] GET OPTION in AGI |
ASTERISK-02821: [patch] See any playing files when STREAM FILE |
ASTERISK-02822: have sip not consider port when matching a peer |
ASTERISK-02823: [patch] Added brazilian tones to zonedata.c |
ASTERISK-02824: [patch] Call progress detection for Brazil |
ASTERISK-02825: SIP channel rings after answer on some calls from Polycom |
ASTERISK-02826: [request + patch?] SIP REGISTER timeout setting wanted |
ASTERISK-02827: sip invite retry with wrong password |
ASTERISK-02828: asterisk UAC function need to send 487 after get CANCEL in sip |
ASTERISK-02829: Dial fails |
ASTERISK-02830: [patch] app_dial use of dial options disallows unlimited timeout without warning. |
ASTERISK-02831: [patch] Support MGCP distinctive ring |
ASTERISK-02832: Retry poll() on EINTR in ast_waitfor_nandfds() |
ASTERISK-02833: [patch] fix silence detection in app_record |
ASTERISK-02834: CVS 1-0 and 1-0-2 mismatch (libpri features/function) |
ASTERISK-02835: Connecting queued person to agent fails |
ASTERISK-02836: [request] Proper pattern matching when using chan_phone |
ASTERISK-02837: DTMF Stops working in Voicemail |
ASTERISK-02838: [patch] chan_mgcp is not functional in CVS HEAD |
ASTERISK-02839: [patch] fix UK CallerID on outward FXS port |
ASTERISK-02840: [patch] [v1-0] Backport locking changes from CVS HEAD + formating/spacing |
ASTERISK-02841: segfault if head caller times out of queue with moh running |
ASTERISK-02842: [PATCH] SIP header continuation line parsing not conformant to RFC |
ASTERISK-02843: [patch] Mailbox getting lost with realtime config |
ASTERISK-02844: [patch] Add a continue option to WaitExten |
ASTERISK-02845: 'i' extension doesnt match invalid dialled extensions, only menu responses |
ASTERISK-02846: Dropping voice to exceptionally long queue |
ASTERISK-02847: if using complex codec and inband dtmf, ast_dsp_process reports wrong codec in error message |
ASTERISK-02848: Use counter not decremented after timeout in INVITE |
ASTERISK-02849: [patch] Misc fixes |
ASTERISK-02850: dsp.c hangup detection on x100p |
ASTERISK-02851: Request for informed comment on behaviour of wctdm driver |
ASTERISK-02852: [patch] Asterisk sends repeated rtp seq number with rfc2833 dtmf |
ASTERISK-02853: I get one way audio when calling from iaxy(ULAW) to zaptel(ALAW) |
ASTERISK-02854: [patch] RTP keepalive for SIP |
ASTERISK-02855: [patch] Improve app_dial M(macro) stuff |
ASTERISK-02856: [patch] leading + and - in Goto/GotoIf to go with priority n stuff |
ASTERISK-02857: app_while |
ASTERISK-02858: [patch] "make samples" always overwrites current configuration |
ASTERISK-02859: V23 CID hangs on polarity reversal that is not a RING |
ASTERISK-02860: Incorrect parsing of remote-party-ID |
ASTERISK-02861: [patch] Multi-Line Comments In Config Files |
ASTERISK-02862: [patch] MacroIf() |
ASTERISK-02863: [patch] Override native transfers by a dialplan predefined extension |
ASTERISK-02864: Call in queue tries twice with 'n' option |
ASTERISK-02865: [patch] Reset variables at SIP reload |
ASTERISK-02866: [patch] Re-use registration credentials |
ASTERISK-02867: SIP to H323 Bridge Issue |
ASTERISK-02868: [patch] iax peers are always reported as lagged on asterisk manager |
ASTERISK-02869: FXO interface stops handling outgoing calls following polarity reversal when using V23 signalling |
ASTERISK-02870: Remove the new line in app_alarmreceiver.c |
ASTERISK-02871: [patch] Supports filaname with a dot in app_record.c |
ASTERISK-02872: [patch] Incoming SIP "Distinctive Ring" detection using Alert-Info header |
ASTERISK-02873: NAT on REGISTER |
ASTERISK-02874: [patch] var_val in res_config_odbc.c |
ASTERISK-02875: [patch] allow zapscan to scan based on channel GROUP |
ASTERISK-02876: [patch] do not increment ts value on ast_rtp_senddigit |
ASTERISK-02877: [patch] Reload dynamic queue members on restart |
ASTERISK-02878: [new_app] WaitIVR |
ASTERISK-02879: contact header patch |
ASTERISK-02880: endless loop due to ast_search_dns() taking too long |
ASTERISK-02881: [request] Implement T.38 as a codec |
ASTERISK-02882: [patch] Let Background() take mutiple files |
ASTERISK-02883: Asterisk does not register with SIP proxies unless a "sip reload" command is issued |
ASTERISK-02884: [patch] fix MEETMESECS in app_meetme |
ASTERISK-02885: chan_mgcp does not send correct ip in sdp until reloaded |
ASTERISK-02886: [patch] SIP channel goes ZOMBIE when transfering call on IAX to MeetMe |
ASTERISK-02887: Non-existant channel type to chan_features crashes Asterisk. |
ASTERISK-02888: [patch] fix memory leak in cdr_odbc.c |
ASTERISK-02889: Previous locking improvements in chan_mgcp need to be revised (???) |
ASTERISK-02890: [patch] Call forwarding to self with chan_sip causes loop. |
ASTERISK-02891: IAX2 transfers fails when one party is behind nat and iax port isn't dnat'ed inside |
ASTERISK-02892: [patch] missing ast_destroy(cfg) |
ASTERISK-02893: [patch] when disallow=all and allow=all in [general], all other codec settings in peers do not workand no sound heard. |
ASTERISK-02894: app_runexten - Run a give extension from CLI |
ASTERISK-02895: [patch] Run script upon registration chan_sip.c |
ASTERISK-02896: [patch] compact sip headers |
ASTERISK-02897: Repeated Zaptel Logger Entry |
ASTERISK-02898: Did the new cvs update break oh323??? |
ASTERISK-02899: [patch] make SQL table name for "cdr" table configurable in cdr_pgsql |
ASTERISK-02900: [patch] Meetme Conference Cloaking and Status Display |
ASTERISK-02901: EAGI fails somehow when a long distance caller connects |
ASTERISK-02902: make progdocs requires graphviz |
ASTERISK-02903: [patch] press a key, record a call. Configured via dialplan variables. |
ASTERISK-02904: E1 PRI won't bring b-channels up |
ASTERISK-02905: canceled voicemail emails corrupt wav |
ASTERISK-02906: [patch] Specify the machine in the safe_asterisk |
ASTERISK-02907: CVS Head doesnt compile! |
ASTERISK-02908: Suspend in tcsh, kill %1 *twice*, then fg gives core dump |
ASTERISK-02909: [patch] print the number of applications registered |
ASTERISK-02910: [PATCH] cdr_dumper_php |
ASTERISK-02911: festival source-code patching not needed anymore |
ASTERISK-02912: SIP response code handling |
ASTERISK-02913: [patch] app_record with silence detection gets read format stuck |
ASTERISK-02914: [patch] Zaptel does not detect hangup in Singapore |
ASTERISK-02915: [patch] Add counters in the show dialplan |
ASTERISK-02916: pbx_realtime.c - Docs and Minor Tweaks |
ASTERISK-02917: Early hangup during MGCP transfer (like unattended transfer) crashes running asterisk |
ASTERISK-02918: [patch] Control codec priorities in IAX2 |
ASTERISK-02919: no tabs for show channels concise |
ASTERISK-02920: [patch] new version of app_sms.c |
ASTERISK-02921: sip.conf parsing error |
ASTERISK-02922: chan_zap.c always tries to use channel 24 as D-channel |
ASTERISK-02923: Recent multiline comment if conf files conflicts with sample sip.conf |
ASTERISK-02924: [patch] H323 channel: in callback send_digit() argument call_token missing |
ASTERISK-02925: Pbx_realtime - SQL Fetch error! |
ASTERISK-02926: MWI system is currently polling based, this makes it event based and fixed ODBC storage issues |
ASTERISK-02927: ALERT_INFO and VXML_URL not sent |
ASTERISK-02928: [PATCH] Send DLCX to MGCP gateway when channel is already dead in Asterisk |
ASTERISK-02929: Strange errors on incoming call |
ASTERISK-02930: [patch] Realtime extensions not resolving variables |
ASTERISK-02931: app_dial leaks frames in some error conditions |
ASTERISK-02932: [patch] CLI 'database showkey <keytree>' |
ASTERISK-02933: [patch] adds shortcutting when selecting translation paths |
ASTERISK-02934: will not load due to strdupa call added to chan_sip.c |
ASTERISK-02935: app_dial DIALSTATUS is always CHANUNAVAIL when not answered |
ASTERISK-02936: Memory leak in ast_expr.y |
ASTERISK-02937: Reload from CLI while using MeetMe causes crash |
ASTERISK-02938: [patch] Hungarian tones (zaptel+indications) |
ASTERISK-02939: using outgoing spool to CONSOLE/dsp and agi sends hangup to other channel |
ASTERISK-02940: faxdetect fails on Asterisk CVS-HEAD-12/07/04-21:50:06 |
ASTERISK-02941: [patch] Patch to allow the directory to exit when user presses "0". |
ASTERISK-02942: app_queue and chan_local maximum usage? |
ASTERISK-02943: [patch] noload'able res_musiconhold.so |
ASTERISK-02944: fmtp payload header |
ASTERISK-02945: chan_agent.so and chan_local.so use count |
ASTERISK-02946: Crash on chan_h323 |
ASTERISK-02947: MusicOnHold and Transferring a call does not work in a queue |
ASTERISK-02948: Asterisk crashes as soon as there are more than 10 calls on the TE110P |
ASTERISK-02949: [patch] MWI subscribe error when type=friend |
ASTERISK-02950: [request] Console/dsp extension does not work with AC97 (48kHz stereo-only) audio chips |
ASTERISK-02951: New batch of phrases for Allison |
ASTERISK-02952: IAX2 to IAX2 huge delays |
ASTERISK-02953: [patch] Fix memory leak in ast_expr.y |
ASTERISK-02954: [patch] res_config_mysql.c causes core dump |
ASTERISK-02955: Using SetVar Incorrectly Causes Asterisk Process to Crash |
ASTERISK-02956: Channel in strange state after transferring 2nd call |
ASTERISK-02957: Can meetme and voice mail integrate |
ASTERISK-02958: [patch] app_read Addition |
ASTERISK-02959: bad clicking/popping only with SMP kernel 2.6.x |
ASTERISK-02960: Report IAX2 Frame_Text to send Hangup Cause from Zap channel |
ASTERISK-02961: [patch] Polish tones (zaptel & asterisk indications) |
ASTERISK-02962: adding SIPURI to predefined channel variables |
ASTERISK-02963: no audio |
ASTERISK-02964: Umm..whats with all the ODBC stuff inside app_voicemail? |
ASTERISK-02965: [PATCH] allow contexts to be repeated in multiple files, so they add together |
ASTERISK-02966: GR-303 Compatibility issues with AFC AccessMAX / UMC1000 |
ASTERISK-02967: allow "category" to be assigned to messages |
ASTERISK-02968: Changes to language handling/minor bug |
ASTERISK-02969: [PATCH] remove need for voicemail-related symlinks in sounds directory |
ASTERISK-02970: utils/smsq.c compile errors on FreeBSD 4.10 |
ASTERISK-02971: Timeout problem with CallerID via DTMF/Polarity on TDM400P |
ASTERISK-02972: [patch] move process_quotes_and_slashes into pbx.c rename to ast_process_quotes_and_slashes |
ASTERISK-02973: [patch] cdr_pgsql.c this has been bugging me. |
ASTERISK-02974: [patch] ODBC Realtime extensions switch requires MySQL |
ASTERISK-02975: [patch] check the RTP version for find invalid frame |
ASTERISK-02976: [patch] Channels get stuck in the parking lot |
ASTERISK-02977: [patch] Send Q.931 cause codes |
ASTERISK-02978: autofallthrough is incompatible with IVR menus; should default to "no" |
ASTERISK-02979: [patch] Move ast_app_has_voicemail and ast_app_messagecount to app_voicemail |
ASTERISK-02980: [patch] Stop MOH on hangup rather than after the channel is destroyed |
ASTERISK-02981: [patch] Clean up a couple of things left behind by "make clean" |
ASTERISK-02982: wrong subroutine name in chan_zap.c |
ASTERISK-02983: [patch] queue priority (weight) |
ASTERISK-02984: res_config_mysql.c Crashes BADLY |
ASTERISK-02985: Crash on masquerading <ZOMBIE> channels |
ASTERISK-02986: Cisco's RTP codec type 100 |
ASTERISK-02987: Read() exists non-zero on timeout |
ASTERISK-02988: Agent Hangup Crashes Asterisk |
ASTERISK-02989: macros cannot exit on */# keys like regular contexts |
ASTERISK-02990: [PATCH] add MacroExit application |
ASTERISK-02991: [patch] Make ast_test_flag and friends a macro |
ASTERISK-02992: [patch] Wrong flags passed to glob, and fix for solaris |
ASTERISK-02993: [patch] Correction to markster's tweak for quad_t |
ASTERISK-02994: [patch] utils/Makefile doesn't work on Solaris |
ASTERISK-02995: mkdep script missing |
ASTERISK-02996: glob call in config.c causes seg fault |
ASTERISK-02997: cdr_csv lets write when cdr_odbc loses connection with a remote database.... |
ASTERISK-02998: RealTime + regexten does not work |
ASTERISK-02999: Fails to compile on Linux FC3 (latest CVS) |